- 5ca7ffd Arguments need to be separated when implementing gyp-actions. by henrike@webrtc.org · 11 years ago
- 7183bbb Cleanup WebRTC tracing by hclam@chromium.org · 11 years ago
- 80fa00b Added modules_unittests.isolate for ndk-apk builds. by henrike@webrtc.org · 11 years ago
- 6f44ab3 Disables unit tests that don't work on Android for Android. by henrike@webrtc.org · 11 years ago
- 5b871f8 Fixes build breakage when building WebRTC in Chromium and having include_tests=1. by henrike@webrtc.org · 11 years ago
- 13efe02 Fixes broken gyp-condition. by henrike@webrtc.org · 11 years ago
- 0642536 Unreverts revert: Makes it possible to find files used by some unit tests when running them as Chrome native tests. by henrike@webrtc.org · 11 years ago
- b7f287d Use scoped_ptr<> for loopback.cc by pbos@webrtc.org · 11 years ago
- a430fef Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the by stefan@webrtc.org · 11 years ago
- ad63306 Revert 4298 "Makes it possible to find files used by some unit t..." by pbos@webrtc.org · 11 years ago
- c82b35c Makes it possible to find files used by some unit tests when running them as Chrome native tests. by henrike@webrtc.org · 11 years ago
- 1d06d1a Adding Stefan as VideoEngine owner, removing Per. by mflodman@webrtc.org · 11 years ago
- e590835 In AudioDeviceWindowsCore::_EnumerateEndpointDevicesAll(), continue enumerating if one individual device failed. by braveyao@webrtc.org · 11 years ago
- 6504a1d Makes it possible to build ndk-apks of native unit tests if the workspace is inside a chromium checkout. by henrike@webrtc.org · 11 years ago
- 5ab7b93 Proper spacing for end-of-namespace comments. by pbos@webrtc.org · 11 years ago
- a539d8e In call to Opus decoder: frame length too large by tina.legrand@webrtc.org · 11 years ago
- 817d63c Possible divide by 0 in ACM. by tina.legrand@webrtc.org · 11 years ago
- 3b7be22 Error in update of read index in ACM by tina.legrand@webrtc.org · 11 years ago
- eaf7428 Rename unit_test.{cc,h} under module_unittest. by pbos@webrtc.org · 11 years ago
- 363852e Remove log of undefined input values in GetCodec. by pbos@webrtc.org · 11 years ago
- 787640d Diff NTP and internal once in VideoCaptureImpl. by pbos@webrtc.org · 11 years ago
- f808b77 Build all java files into jar for each module on Android by fischman@webrtc.org · 11 years ago
- 5934a6a WebRTCViEDemo: Use global reference when passing variables across different threads by yujie.mao@webrtc.org · 11 years ago
- cdfff5b Android opengles renderer: add thread sync to swap frame and draw native. by braveyao@webrtc.org · 11 years ago
- c9ba795 Suppress excessive logging in video_coding by hclam@chromium.org · 11 years ago
- 0c9b40d Moves tools/update.py to trunk/webrtc/tools and updates it so that it no longer pulls any information from the DEPS file. by henrike@webrtc.org · 11 years ago
- 399baf7 Removes unused main function that is poluting the build. by henrike@webrtc.org · 11 years ago
- 040e75f Re-add WebRTCDemo dependencies as dependencies (not just inputs) because they also need to be built for this target! by fischman@webrtc.org · 11 years ago
- 1d9d1ea Move TickTime::QueryOsForTicks out-of-line by fischman@webrtc.org · 11 years ago
- f4a9648 Removes kStateFree and kStateDecoding, added a free_frames_ list which simplifies finding a free frame. by stefan@webrtc.org · 11 years ago
- 9f56b60 Fixed bad parameter passing in compare_videos.py by phoglund@webrtc.org · 11 years ago
- 4171693 Fix unnamed-type-template-args warnings on clang. by pbos@webrtc.org · 11 years ago
- d32fe69 Correctly rebuild WebRTCDemo-debug.apk after modules/ source file changes. by fischman@webrtc.org · 11 years ago
- 9165c4d Adding a first simple version of overuse detection, but not hooked up. by mflodman@webrtc.org · 11 years ago
- 7401259 Removed ViE file API. by mflodman@webrtc.org · 11 years ago
- f4ac411 Do basic parsing of RTCP headers in PcapFileReader to enable log filtering. by solenberg@webrtc.org · 11 years ago
- 281399a Remove unused multi stream bandwidth estimator. by solenberg@webrtc.org · 11 years ago
- a1e06c2 Make sure padding packets are sent. by stefan@webrtc.org · 11 years ago
- 6cf9867 mac: Mark kCGLPFAFullScreen as allowed in a 10.6-only function. by sergeyu@chromium.org · 11 years ago
- 381c0a0 Fix memory bot failure by hclam@chromium.org · 11 years ago
- 19f7ac1 Enqueue packet in pacer if sending fails by hclam@chromium.org · 11 years ago
- b211732 VCM: removing max jitter estimate by mikhal@webrtc.org · 11 years ago
- f4e4324 Remove unneeded *_NOT_SUPPORTED from VoEAudioProcessing. by andrew@webrtc.org · 11 years ago
- 0d2e502 Fixes some pacer/padding issues found while testing. by stefan@webrtc.org · 11 years ago
- 350c135 Use 3 threads for higher than 720p resolutions by fbarchard@google.com · 11 years ago
- 6b7e468 Add a log message to see video delay break down by hclam@chromium.org · 11 years ago
- 037b232 Make ScreenCapturerMac work in versions of OSX before Lion. by sergeyu@chromium.org · 11 years ago
- b06153b Enable ScreenCapturer unittests by sergeyu@chromium.org · 11 years ago
- d85ab45 Use intptr_t to represent window IDs on all platforms. by sergeyu@chromium.org · 11 years ago
- 39f2547 Wire up pacer-based padding. by stefan@webrtc.org · 11 years ago
- 3777209 Revert r4145 "Revert 4127 "Switch frame list implementation to std::map."" by stefan@webrtc.org · 11 years ago
- 81751f0 Revert r4146 "Revert 4104 "Refactor jitter buffer to use separate lists for de..."" by stefan@webrtc.org · 11 years ago
- 2eb0c7a Fix AV sync issue by hclam@chromium.org · 11 years ago
- eaeb84f Log current and target AV delay in ViESyncModule by hclam@chromium.org · 11 years ago
- e797d9e Merge more tests into modules_{unit,integration}tests. by kjellander@webrtc.org · 11 years ago
- d6f0906 WebRTCDemo: ensures that using front and back camera work as expected. by henrike@webrtc.org · 11 years ago
- 798d5c1 Fixes linker issue with no op trace. by henrike@webrtc.org · 11 years ago
- 92717b6 Risk of division by zero. by turaj@webrtc.org · 11 years ago
- 9fefc91 Revert 4211 "Build all java files into jar for each module on An..." by fischman@webrtc.org · 11 years ago
- 3b37c8a Disable ViEExtendedIntegrationTest.RunsCodecTestWithoutErrors test. by kjellander@webrtc.org · 11 years ago
- 936844c Disable ViEExtendedIntegrationTest.RunsCodecTestWithoutErrors test. by kjellander@webrtc.org · 11 years ago
- 74819dd Fix breakage due to test_fec conversion to gtest. by kjellander@webrtc.org · 11 years ago
- 86fb841 Convert test_fec to gtest by kjellander@webrtc.org · 11 years ago
- af38b53 Disable ViEStandardIntegrationTest.RunsRtpRtcpTestWithoutErrors test. by kjellander@webrtc.org · 11 years ago
- d6ac5c3 G722_1/G722_1C codecs won't instantiate by tina.legrand@webrtc.org · 11 years ago
- 1df6cc7 Reorganize test targets in WebRTC by kjellander@webrtc.org · 11 years ago
- f92d9ad Build all java files into jar for each module on Android by fischman@webrtc.org · 11 years ago
- 3da595e Allow the screen capturer to capture oversized cursors and cursors without alpha channel (Windows). by alexeypa@chromium.org · 11 years ago
- a83e538 Landing binary cursor image files to be used in a follow up CL. by alexeypa@chromium.org · 11 years ago
- ccc21d2 Updated WebRTC version to 3.33 by elham@webrtc.org · 11 years ago
- 091c4f8 Making no NACK mode work again in VideoEngine. by mflodman@webrtc.org · 11 years ago
- 7556bbe RW lock access to ssrc maps in VideoCall. by pbos@webrtc.org · 11 years ago
- a6e8ec3 Add back the WEBRTC_DIRECT_TRACE flag. by solenberg@webrtc.org · 11 years ago
- ffce2b1 AudioDeviceAndroidOpenSLES: NULL variables might be referenced in StopPlayout() by braveyao@webrtc.org · 11 years ago
- 6e5b871 Revert some variables to uint32_t to fix compile errors on Mac gcc. by andrew@webrtc.org · 11 years ago
- 4477bd5 Allow audio devices with up to 64 channels on Mac. by andrew@webrtc.org · 11 years ago
- b42bf4b Fixed Rtp/Rtcp tests by pwestin@webrtc.org · 11 years ago
- 9c49814 Fix relative path to .gitignore and other minor changes. by andrew@webrtc.org · 11 years ago
- a052973 Removing functionality for inserting pre-encoded frames instead of raw by mflodman@webrtc.org · 11 years ago
- 306e331 Add script for appending entries to .gitignore. by andrew@webrtc.org · 11 years ago
- 4dca856 Fix size_t to int conversion error on Win64. by andrew@webrtc.org · 11 years ago
- 1632b97 Remove fake screen capturer because it's not used anywhere. by sergeyu@chromium.org · 11 years ago
- f22cc80 Fix for STL vector function data not available. by pwestin@webrtc.org · 11 years ago
- f94ddea Connect ACM with RTP module for audio NACK. by pwestin@webrtc.org · 11 years ago
- a629133 Nack for audio. by turaj@webrtc.org · 11 years ago
- a1e84f1 Fix leaks in DesktopRegion by sergeyu@chromium.org · 11 years ago
- 018870d Implement DetectNumberOfCores on Android and make it consistent on Linux and Android by fischman@webrtc.org · 11 years ago
- 4673a99 Wire up Nack for Voe by pwestin@webrtc.org · 11 years ago
- 767ca95 Fix init list for VideoSendStream::Config::Rtp. by pbos@webrtc.org · 11 years ago
- 2c343fc Stats+Config moved into VideoSend/ReceiveStreams. by pbos@webrtc.org · 11 years ago
- 561fe8b Merge webrtc_utility_unittests into modules_unittests. by kjellander@webrtc.org · 11 years ago
- 14035f1 Restore relative include paths to libyuv. by andrew@webrtc.org · 11 years ago
- fd5d808 Issue 1847, memcopy is wrong and unnecessary, it is sufficient to store the pointer before clearing the instance, and write back the pointer. by turaj@webrtc.org · 11 years ago
- 07e10ab resolve b9050210. Avoid pushing sync packet before any packet received. Do not turn on AV-sync if initial delay is zero. by turaj@webrtc.org · 11 years ago
- 910a3c6 Move screen capturers from chromium to webrtc. by sergeyu@chromium.org · 11 years ago
- 71fe9ac Refactor padding and rtp header functionality. by stefan@webrtc.org · 11 years ago
- 4dc727b Update the remote bitrate estimator before passing the packet to the RTP module. by stefan@webrtc.org · 11 years ago
- 84119ff Remove XvRenderer. by pbos@webrtc.org · 11 years ago
- 5eb0408 Fix build error introduced with r4168. by stefan@webrtc.org · 11 years ago
- 39278fb Add support for padding in pacer. by stefan@webrtc.org · 11 years ago